So in conclusion, its all about file size. Neato :thumbsup:
Of course you can train yourself to hear the difference between compressed and uncompressed audio. Compression leaves artifacts that you can hear if you know what you're listening for. Well done compression does it in such a way that its not obvious, but its still there.
For comparison, Some people can't spot the difference between a JPG image and a TGA. If you know what artifacts are likely to occur and where to look, its reasonably easy to spot these.
For audio compression I ran ABX testing on OGG samples to determine what I should compress my CD library with. Ended up about 350kb or so was where I started having a really hard time telling the difference. So, I went slightly higher than that and compessed it all.
In both cases, its not about being physically unable to hear or see the compression. Its about knowing what to look for.
If you throw enough bits at lossy compression, it eventually becomes lossless. Lossy compression gives you the option to determine where it is that you can hear the difference.
It is just silly to say that you can ALWAYS tell the difference between lossy and lossless even if you know what you are looking for.
BTW, I am interested, when did you run this ABX test? 350kb/s for OGG (currently) is really quite high. Also, What was your confidence level?
Depends on where you are at. If you are on the highway with roadnoise? No.
In the room I posted on page 2? yes.
Even in the room you posted. With the absolute best sound system money can buy, I can raise the bitrate to a high enough level that the human ear can't tell the difference.
Heck, I can raise the bitrate high enough that no measuring device can detect the difference, because the output will be exactly identical.
Like I said before, if you raise the bitrate high enough on any lossy codec, you end up with a lossless stream.
If you throw enough bits at lossy compression, it eventually becomes lossless.
If you throw enough bits at lossy compression, it eventually becomes lossless. Lossy compression gives you the option to determine where it is that you can hear the difference.
It is just silly to say that you can ALWAYS tell the difference between lossy and lossless even if you know what you are looking for.
BTW, I am interested, when did you run this ABX test? 350kb/s for OGG (currently) is really quite high. Also, What was your confidence level?
That's not true of lossy compression schemes in general. For example, youn't use MP3 or JPEG encoding as a lossless compression format regardless of the bit rate or compression parameters you use.
err, no. Lossless compression is reversible. Lossy compression never is.
See my explanation above. Most all lossy compression is based off of algorithms which can easily be turned into lossless compression.
Lossy compression operates on the premise of dropping parts of the waveform that you might not notice. But the waveform is changed for sure. Doesn't matter how little compression you use, you are still dropping parts of the waveform.
Take a wave file, plot the waveform, compress, then compare the waveform to the original wave file and you'll see differences. Not so with lossless compression.
If you throw enough bits at lossy compression, it eventually becomes lossless. Lossy compression gives you the option to determine where it is that you can hear the difference.
It is just silly to say that you can ALWAYS tell the difference between lossy and lossless even if you know what you are looking for.
BTW, I am interested, when did you run this ABX test? 350kb/s for OGG (currently) is really quite high. Also, What was your confidence level?
Just because it isn't generally done doesn't mean that it isn't possible.
The lossy compression comes in because the compressors are throwing away data from the DCT. Raising the bitrate high enough would essentially be saying "Ok, keep everything in the DCT." which is a lossless compression scheme.
It's not possible in the context of the discussion we're having now. You can't encode a arbitrary song as an MP3 at bitrate that would make it lossless. The format simply doesn't support that.
You're assuming the compression format allows encoding an arbitrary DCT without throwing away data. They don't.
True, there is a specification maximum bitrate, however, that is more a "Nobody will ever need this" sort of decision then a "It can't be done" decision.
MP3 has this limitation, AAC does not.
There is some possible (not definite) data loss due to rounding as well.
No, you really don't understand the problem here. When transforming a 16-bit PCM audio stream using some DCT algorithim you get a sequence of floating point values. If you then encoded these floating point values with enough precision and range, then yes you could feed them into the inverse DCT algorithm and get the original 16-bit PCM values back. However, the MP3 and AAC encoding standards do not let you do this.
These standards require that the (M)DCT samples be quantized and then Huffman encoded before they're output. The quantization step is crucially important, this takes the DCT samples and turns them into something that can be compressed well by the Huffman encoder. This quantization step however is what makes the MP3, AAC and another similar compression schemes lossy, and what makes them unavoidably lossy. MP3 requires that quantization produce an integer in the range of 8206 and 8206 (AAC between -8191 and 8192). That's little over 14 bits, and not any where near enough bits to allow the inverse DCT to reconstruct the original 16-bit PCM values in the general or typical case.
No, you really don't understand the problem here. When transforming a 16-bit PCM audio stream using some DCT algorithim you get a sequence of floating point values. If you then encoded these floating point values with enough precision and range, then yes you could feed them into the inverse DCT algorithm and get the original 16-bit PCM values back. However, the MP3 and AAC encoding standards do not let you do this.
These standards require that the (M)DCT samples be quantized and then Huffman encoded before they're output. The quantization step is crucially important, this takes the DCT samples and turns them into something that can be compressed well by the Huffman encoder. This quantization step however is what makes the MP3, AAC and another similar compression schemes lossy, and what makes them unavoidably lossy. MP3 requires that quantization produce an integer in the range of 8206 and 8206 (AAC between -8191 and 8192). That's little over 14 bits, and not any where near enough bits to allow the inverse DCT to reconstruct the original 16-bit PCM values in the general or typical case.